前言

注意这个需要对端发起呼叫，调试的这端作为被呼叫方。

SetCertificate里面是个不断循环的，直到有answer请求的时候，才会继续往下走。

最终生成的就是jsepSessionDescription对象，有了这个信息再转为sdp，传回给呼叫方。呼叫方收到后调用SetRemoteDescription就完成了媒体协商。

PeerConnection::Create
PeerConnection::Initialize
SdpOfferAnswerHandler::Create
SdpOfferAnswerHandler::Initialize
-》
std::makeunique(
signaling_thread(), channel_manager(), this, pc->sessionid(),
pc->dtlsenabled(), std::move(dependencies.cert_generator),
certificate, &ssrc_generator,
this {
transport_controller()->SetLocalCertificate(certificate);
})

WebRtcSessionDescriptionFactory::WebRtcSessionDescriptionFactory

WebRtcSessionDescriptionFactory::WebRtcSessionDescriptionFactory(
    rtc::Thread* signaling_thread,
    cricket::ChannelManager* channel_manager,
    const SdpStateProvider* sdp_info,
    const std::string& session_id,
    bool dtls_enabled,
    std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
    const rtc::scoped_refptr<rtc::RTCCertificate>& certificate,
    UniqueRandomIdGenerator* ssrc_generator,
    std::function<void(const rtc::scoped_refptr<rtc::RTCCertificate>&)>
        on_certificate_ready)
    : signaling_thread_(signaling_thread),
      session_desc_factory_(channel_manager,
                            &transport_desc_factory_,
                            ssrc_generator),
      // RFC 4566 suggested a Network Time Protocol (NTP) format timestamp
      // as the session id and session version. To simplify, it should be fine
      // to just use a random number as session id and start version from
      // |kInitSessionVersion|.
      session_version_(kInitSessionVersion),
      cert_generator_(dtls_enabled ? std::move(cert_generator) : nullptr),
      sdp_info_(sdp_info),
      session_id_(session_id),
      certificate_request_state_(CERTIFICATE_NOT_NEEDED),
      on_certificate_ready_(on_certificate_ready) {
  RTC_DCHECK(signaling_thread_);
  if (!dtls_enabled) {
    SetSdesPolicy(cricket::SEC_REQUIRED);
    RTC_LOG(LS_VERBOSE) << "DTLS-SRTP disabled.";
    return;
  }
  // SRTP-SDES is disabled if DTLS is on.
  SetSdesPolicy(cricket::SEC_DISABLED);
  if (certificate) {
    // Use |certificate|.
    certificate_request_state_ = CERTIFICATE_WAITING;
    RTC_LOG(LS_VERBOSE) << "DTLS-SRTP enabled; has certificate parameter.";
    // We already have a certificate but we wait to do |SetIdentity|; if we do
    // it in the constructor then the caller has not had a chance to connect to
    // |SignalCertificateReady|.
    signaling_thread_->Post(
        RTC_FROM_HERE, this, MSG_USE_CONSTRUCTOR_CERTIFICATE,
        new rtc::ScopedRefMessageData<rtc::RTCCertificate>(certificate));
  } else { //创建证书
    // Generate certificate.
    certificate_request_state_ = CERTIFICATE_WAITING; //设置状态
      //创建回调绑定
    rtc::scoped_refptr<WebRtcCertificateGeneratorCallback> callback(
        new rtc::RefCountedObject<WebRtcCertificateGeneratorCallback>());
    callback->SignalRequestFailed.connect(
        this, &WebRtcSessionDescriptionFactory::OnCertificateRequestFailed);
    callback->SignalCertificateReady.connect(
        this, &WebRtcSessionDescriptionFactory::SetCertificate); 
    rtc::KeyParams key_params = rtc::KeyParams();
    RTC_LOG(LS_VERBOSE)
        << "DTLS-SRTP enabled; sending DTLS identity request (key type: "
        << key_params.type() << ").";
      // 异步创建证书
    // Request certificate. This happens asynchronously, so that the caller gets
    // a chance to connect to |SignalCertificateReady|.
    cert_generator_->GenerateCertificateAsync(key_params, absl::nullopt,
                                              callback);
  }
}

cert_generator_->GenerateCertificateAsync(key_params, absl::nullopt,<br />                                              callback);

RTCCertificateGenerator::GenerateCertificateAsync

void RTCCertificateGenerator::GenerateCertificateAsync(
    const KeyParams& key_params,
    const absl::optional<uint64_t>& expires_ms,
    const scoped_refptr<RTCCertificateGeneratorCallback>& callback) {
  RTC_DCHECK(signaling_thread_->IsCurrent());
  RTC_DCHECK(callback);
  // Create a new |RTCCertificateGenerationTask| for this generation request. It
  // is reference counted and referenced by the message data, ensuring it lives
  // until the task has completed (independent of |RTCCertificateGenerator|).
  worker_thread_->PostTask(RTC_FROM_HERE, [key_params, expires_ms,
                                           signaling_thread = signaling_thread_,
                                           cb = callback]() {
    scoped_refptr<RTCCertificate> certificate =
        RTCCertificateGenerator::GenerateCertificate(key_params, expires_ms);
    signaling_thread->PostTask(
        RTC_FROM_HERE, [cert = std::move(certificate), cb = std::move(cb)]() {
          cert ? cb->OnSuccess(cert) : cb->OnFailure();
        });
  });
}

异步去创建证书，里面主要是workerthread->PostTask，向工作线程发送创建任务

SdpOfferAnswerHandler::CreateAnswer

void SdpOfferAnswerHandler::CreateAnswer(
    CreateSessionDescriptionObserver* observer,
    const PeerConnectionInterface::RTCOfferAnswerOptions& options) {
  RTC_DCHECK_RUN_ON(signaling_thread());
  // Chain this operation. If asynchronous operations are pending on the chain,
  // this operation will be queued to be invoked, otherwise the contents of the
  // lambda will execute immediately.
  operations_chain_->ChainOperation(
      [this_weak_ptr = weak_ptr_factory_.GetWeakPtr(),
       observer_refptr =
           rtc::scoped_refptr<CreateSessionDescriptionObserver>(observer),
       options](std::function<void()> operations_chain_callback) {
        // Abort early if |this_weak_ptr| is no longer valid.
        if (!this_weak_ptr) {
          observer_refptr->OnFailure(RTCError(
              RTCErrorType::INTERNAL_ERROR,
              "CreateAnswer failed because the session was shut down"));
          operations_chain_callback();
          return;
        }
        // The operation completes asynchronously when the wrapper is invoked.
        rtc::scoped_refptr<CreateSessionDescriptionObserverOperationWrapper>
            observer_wrapper(new rtc::RefCountedObject<
                             CreateSessionDescriptionObserverOperationWrapper>(
                std::move(observer_refptr),
                std::move(operations_chain_callback)));
        this_weak_ptr->DoCreateAnswer(options, observer_wrapper);
      });
}

后面主要调用DoCreateAnswer

SdpOfferAnswerHandler::DoCreateAnswer

void SdpOfferAnswerHandler::DoCreateAnswer(
    const PeerConnectionInterface::RTCOfferAnswerOptions& options,
    rtc::scoped_refptr<CreateSessionDescriptionObserver> observer) {
  RTC_DCHECK_RUN_ON(signaling_thread());
  TRACE_EVENT0("webrtc", "SdpOfferAnswerHandler::DoCreateAnswer");
  if (!observer) {
    RTC_LOG(LS_ERROR) << "CreateAnswer - observer is NULL.";
    return;
  }
  // If a session error has occurred the PeerConnection is in a possibly
  // inconsistent state so fail right away.
  if (session_error() != SessionError::kNone) {
    std::string error_message = GetSessionErrorMsg();
    RTC_LOG(LS_ERROR) << "CreateAnswer: " << error_message;
    pc_->message_handler()->PostCreateSessionDescriptionFailure(
        observer,
        RTCError(RTCErrorType::INTERNAL_ERROR, std::move(error_message)));
    return;
  }
  if (!(signaling_state_ == PeerConnectionInterface::kHaveRemoteOffer ||
        signaling_state_ == PeerConnectionInterface::kHaveLocalPrAnswer)) {
    std::string error =
        "PeerConnection cannot create an answer in a state other than "
        "have-remote-offer or have-local-pranswer.";
    RTC_LOG(LS_ERROR) << error;
    pc_->message_handler()->PostCreateSessionDescriptionFailure(
        observer, RTCError(RTCErrorType::INVALID_STATE, std::move(error)));
    return;
  }
  // The remote description should be set if we're in the right state.
  RTC_DCHECK(remote_description());
  if (IsUnifiedPlan()) {
    if (options.offer_to_receive_audio !=
        PeerConnectionInterface::RTCOfferAnswerOptions::kUndefined) {
      RTC_LOG(LS_WARNING) << "CreateAnswer: offer_to_receive_audio is not "
                             "supported with Unified Plan semantics. Use the "
                             "RtpTransceiver API instead.";
    }
    if (options.offer_to_receive_video !=
        PeerConnectionInterface::RTCOfferAnswerOptions::kUndefined) {
      RTC_LOG(LS_WARNING) << "CreateAnswer: offer_to_receive_video is not "
                             "supported with Unified Plan semantics. Use the "
                             "RtpTransceiver API instead.";
    }
  }
  cricket::MediaSessionOptions session_options;
  GetOptionsForAnswer(options, &session_options);
  webrtc_session_desc_factory_->CreateAnswer(observer, session_options);
}

里面也是主要是后面的 webrtcsession_desc_factory->CreateAnswer(observer, session_options);

WebRtcSessionDescriptionFactory::CreateAnswer

void WebRtcSessionDescriptionFactory::CreateAnswer(
    CreateSessionDescriptionObserver* observer,
    const cricket::MediaSessionOptions& session_options) {
  std::string error = "CreateAnswer";
  if (certificate_request_state_ == CERTIFICATE_FAILED) {
    error += kFailedDueToIdentityFailed;
    RTC_LOG(LS_ERROR) << error;
    PostCreateSessionDescriptionFailed(observer, error);
    return;
  }
  if (!sdp_info_->remote_description()) {
    error += " can't be called before SetRemoteDescription.";
    RTC_LOG(LS_ERROR) << error;
    PostCreateSessionDescriptionFailed(observer, error);
    return;
  }
  if (sdp_info_->remote_description()->GetType() != SdpType::kOffer) {
    error += " failed because remote_description is not an offer.";
    RTC_LOG(LS_ERROR) << error;
    PostCreateSessionDescriptionFailed(observer, error);
    return;
  }
  if (!ValidMediaSessionOptions(session_options)) {
    error += " called with invalid session options.";
    RTC_LOG(LS_ERROR) << error;
    PostCreateSessionDescriptionFailed(observer, error);
    return;
  }
    //创建请求
  CreateSessionDescriptionRequest request(
      CreateSessionDescriptionRequest::kAnswer, observer, session_options);
  if (certificate_request_state_ == CERTIFICATE_WAITING) {
    create_session_description_requests_.push(request);
  } else {
    RTC_DCHECK(certificate_request_state_ == CERTIFICATE_SUCCEEDED ||
               certificate_request_state_ == CERTIFICATE_NOT_NEEDED);
    InternalCreateAnswer(request);
  }
}

请求完成后，就调用 WebRtcCertificateGeneratorCallback::OnSuccess

WebRtcCertificateGeneratorCallback::OnSuccess

这时候就会进来WebRtcSessionDescriptionFactory::SetCertificate

WebRtcSessionDescriptionFactory::SetCertificate

void WebRtcSessionDescriptionFactory::SetCertificate(
    const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) {
  RTC_DCHECK(certificate);
  RTC_LOG(LS_VERBOSE) << "Setting new certificate.";
  certificate_request_state_ = CERTIFICATE_SUCCEEDED;
    //设置信号
  on_certificate_ready_(certificate);
  transport_desc_factory_.set_certificate(certificate);
  transport_desc_factory_.set_secure(cricket::SEC_ENABLED);
  while (!create_session_description_requests_.empty()) {
    if (create_session_description_requests_.front().type ==
        CreateSessionDescriptionRequest::kOffer) {
      InternalCreateOffer(create_session_description_requests_.front());
    } else {
      InternalCreateAnswer(create_session_description_requests_.front());
    }
    create_session_description_requests_.pop();
  }
}

这里现在就会进去到WebRtcSessionDescriptionFactory::InternalCreateAnswer
注意该函数SetCertificate会一直执行，直到有个answer请求才会真正的去创建一个answer。

WebRtcSessionDescriptionFactory::InternalCreateAnswer

void WebRtcSessionDescriptionFactory::InternalCreateAnswer(
    CreateSessionDescriptionRequest request) {
  if (sdp_info_->remote_description()) {
    for (cricket::MediaDescriptionOptions& options :
         request.options.media_description_options) {
      // According to http://tools.ietf.org/html/rfc5245#section-9.2.1.1
      // an answer should also contain new ICE ufrag and password if an offer
      // has been received with new ufrag and password.
      options.transport_options.ice_restart =
          sdp_info_->IceRestartPending(options.mid);
      // We should pass the current DTLS role to the transport description
      // factory, if there is already an existing ongoing session.
      absl::optional<rtc::SSLRole> dtls_role =
          sdp_info_->GetDtlsRole(options.mid);
      if (dtls_role) {
        options.transport_options.prefer_passive_role =
            (rtc::SSL_SERVER == *dtls_role);
      }
    }
  }
  std::unique_ptr<cricket::SessionDescription> desc =
      session_desc_factory_.CreateAnswer(
          sdp_info_->remote_description()
              ? sdp_info_->remote_description()->description()
              : nullptr,
          request.options,
          sdp_info_->local_description()
              ? sdp_info_->local_description()->description()
              : nullptr);
  if (!desc) {
    PostCreateSessionDescriptionFailed(request.observer,
                                       "Failed to initialize the answer.");
    return;
  }
  // RFC 3264
  // If the answer is different from the offer in any way (different IP
  // addresses, ports, etc.), the origin line MUST be different in the answer.
  // In that case, the version number in the "o=" line of the answer is
  // unrelated to the version number in the o line of the offer.
  // Get a new version number by increasing the |session_version_answer_|.
  // The |session_version_| is a uint64_t, the wrap around should not happen.
  RTC_DCHECK(session_version_ + 1 > session_version_);
  auto answer = std::make_unique<JsepSessionDescription>(
      SdpType::kAnswer, std::move(desc), session_id_,
      rtc::ToString(session_version_++));
  if (sdp_info_->local_description()) {
    // Include all local ICE candidates in the SessionDescription unless
    // the remote peer has requested an ICE restart.
    for (const cricket::MediaDescriptionOptions& options :
         request.options.media_description_options) {
      if (!options.transport_options.ice_restart) {
        CopyCandidatesFromSessionDescription(sdp_info_->local_description(),
                                             options.mid, answer.get());
      }
    }
  }
  PostCreateSessionDescriptionSucceeded(request.observer, std::move(answer));
}

主要是
auto answer = std::makeunique(
SdpType::kAnswer, std::move(desc), session_id,
rtc::ToString(sessionversion++));
最终我们要获得就是这个answer。

前面调用了
std::uniqueptr desc =
session_desc_factory.CreateAnswer(
sdpinfo->remotedescription()
? sdp_info->remotedescription()->description()
: nullptr,
request.options,
sdp_info->localdescription()
? sdp_info->local_description()->description()
: nullptr);

MediaSessionDescriptionFactory::CreateAnswer

std::unique_ptr<SessionDescription>
MediaSessionDescriptionFactory::CreateAnswer(
    const SessionDescription* offer,
    const MediaSessionOptions& session_options,
    const SessionDescription* current_description) const {
**
****
// 创建了answer，后面的内容都是往里面填东西
  auto answer = std::make_unique<SessionDescription>();
  // If the offer supports BUNDLE, and we want to use it too, create a BUNDLE
  // group in the answer with the appropriate content names.
  const ContentGroup* offer_bundle = offer->GetGroupByName(GROUP_TYPE_BUNDLE);
  ContentGroup answer_bundle(GROUP_TYPE_BUNDLE);
  // Transport info shared by the bundle group.
  std::unique_ptr<TransportInfo> bundle_transport;
  answer->set_extmap_allow_mixed(offer->extmap_allow_mixed());
  // Iterate through the media description options, matching with existing
  // media descriptions in |current_description|.
  size_t msection_index = 0;
  for (const MediaDescriptionOptions& media_description_options :
       session_options.media_description_options) {
    const ContentInfo* offer_content = &offer->contents()[msection_index];
    // Media types and MIDs must match between the remote offer and the
    // MediaDescriptionOptions.
    RTC_DCHECK(
        IsMediaContentOfType(offer_content, media_description_options.type));
    RTC_DCHECK(media_description_options.mid == offer_content->name);
    const ContentInfo* current_content = nullptr;
    if (current_description &&
        msection_index < current_description->contents().size()) {
      current_content = &current_description->contents()[msection_index];
    }
    RtpHeaderExtensions header_extensions = RtpHeaderExtensionsFromCapabilities(
        UnstoppedRtpHeaderExtensionCapabilities(
            media_description_options.header_extensions));
      // 根据类型来添加内容
    switch (media_description_options.type) {
      case MEDIA_TYPE_AUDIO:
        if (!AddAudioContentForAnswer(
         **
        }
        break;
      case MEDIA_TYPE_VIDEO:
        if (!AddVideoContentForAnswer(
           **
        }
        break;
      case MEDIA_TYPE_DATA:
        if (!AddDataContentForAnswer(
           **
        }
        break;
      case MEDIA_TYPE_UNSUPPORTED:
        if (!AddUnsupportedContentForAnswer(
          **
      default:
        RTC_NOTREACHED();
    }
    ++msection_index;
    // See if we can add the newly generated m= section to the BUNDLE group in
    // the answer.
    ContentInfo& added = answer->contents().back();
    if (!added.rejected && session_options.bundle_enabled && offer_bundle &&
        offer_bundle->HasContentName(added.name)) {
      answer_bundle.AddContentName(added.name);
      bundle_transport.reset(
          new TransportInfo(*answer->GetTransportInfoByName(added.name)));
    }
  }
 ***
  return answer;
}

这里以调用音频的AddAudioContentForAnswer为例子

MediaSessionDescriptionFactory::AddAudioContentForAnswer

// |audio_codecs| = set of all possible codecs that can be used, with correct
// payload type mappings
//
// |supported_audio_codecs| = set of codecs that are supported for the direction
// of this m= section
//
// acd->codecs() = set of previously negotiated codecs for this m= section
//
// The payload types should come from audio_codecs, but the order should come
// from acd->codecs() and then supported_codecs, to ensure that re-offers don't
// change existing codec priority, and that new codecs are added with the right
// priority.
bool MediaSessionDescriptionFactory::AddAudioContentForAnswer(
    const MediaDescriptionOptions& media_description_options,
    const MediaSessionOptions& session_options,
    const ContentInfo* offer_content,
    const SessionDescription* offer_description,
    const ContentInfo* current_content,
    const SessionDescription* current_description,
    const TransportInfo* bundle_transport,
    const AudioCodecs& audio_codecs,
    const RtpHeaderExtensions& default_audio_rtp_header_extensions,
    StreamParamsVec* current_streams,
    SessionDescription* answer,
    IceCredentialsIterator* ice_credentials) const {
  RTC_CHECK(IsMediaContentOfType(offer_content, MEDIA_TYPE_AUDIO));
  const AudioContentDescription* offer_audio_description =
      offer_content->media_description()->as_audio();
    // 创建音频的CreateTransportAnswer
  std::unique_ptr<TransportDescription> audio_transport = CreateTransportAnswer(
      media_description_options.mid, offer_description,
      media_description_options.transport_options, current_description,
      bundle_transport != nullptr, ice_credentials);
  if (!audio_transport) {
    return false;
  }
  // Pick codecs based on the requested communications direction in the offer
  // and the selected direction in the answer.
  // Note these will be filtered one final time in CreateMediaContentAnswer.
  auto wants_rtd = media_description_options.direction;
  auto offer_rtd = offer_audio_description->direction();
  auto answer_rtd = NegotiateRtpTransceiverDirection(offer_rtd, wants_rtd);
  AudioCodecs supported_audio_codecs =
      GetAudioCodecsForAnswer(offer_rtd, answer_rtd);
  AudioCodecs filtered_codecs;
    // 过滤音频编码器
  if (!media_description_options.codec_preferences.empty()) {
    filtered_codecs = MatchCodecPreference(
        media_description_options.codec_preferences, supported_audio_codecs);
  } else {
    // Add the codecs from current content if it exists and is not rejected nor
    // recycled.
    if (current_content && !current_content->rejected &&
        current_content->name == media_description_options.mid) {
      RTC_CHECK(IsMediaContentOfType(current_content, MEDIA_TYPE_AUDIO));
      const AudioContentDescription* acd =
          current_content->media_description()->as_audio();
      for (const AudioCodec& codec : acd->codecs()) {
        if (FindMatchingCodec<AudioCodec>(acd->codecs(), audio_codecs, codec,
                                          nullptr)) {
          filtered_codecs.push_back(codec);
        }
      }
    }
    // Add other supported audio codecs.
    for (const AudioCodec& codec : supported_audio_codecs) {
      if (FindMatchingCodec<AudioCodec>(supported_audio_codecs, audio_codecs,
                                        codec, nullptr) &&
          !FindMatchingCodec<AudioCodec>(supported_audio_codecs,
                                         filtered_codecs, codec, nullptr)) {
        // We should use the local codec with local parameters and the codec id
        // would be correctly mapped in |NegotiateCodecs|.
        filtered_codecs.push_back(codec);
      }
    }
  }
  bool bundle_enabled = offer_description->HasGroup(GROUP_TYPE_BUNDLE) &&
                        session_options.bundle_enabled;
  auto audio_answer = std::make_unique<AudioContentDescription>();
  // Do not require or create SDES cryptos if DTLS is used.
  cricket::SecurePolicy sdes_policy =
      audio_transport->secure() ? cricket::SEC_DISABLED : secure();
  if (!SetCodecsInAnswer(offer_audio_description, filtered_codecs,
                         media_description_options, session_options,
                         ssrc_generator_, current_streams,
                         audio_answer.get())) {
    return false;
  }
    // 构建音频的MediaContent
  if (!CreateMediaContentAnswer(
          offer_audio_description, media_description_options, session_options,
          sdes_policy, GetCryptos(current_content),
          filtered_rtp_header_extensions(default_audio_rtp_header_extensions),
          ssrc_generator_, enable_encrypted_rtp_header_extensions_,
          current_streams, bundle_enabled, audio_answer.get())) {
    return false;  // Fails the session setup.
  }
  bool secure = bundle_transport ? bundle_transport->description.secure()
                                 : audio_transport->secure();
  bool rejected = media_description_options.stopped ||
                  offer_content->rejected ||
                  !IsMediaProtocolSupported(MEDIA_TYPE_AUDIO,
                                            audio_answer->protocol(), secure);
  if (!AddTransportAnswer(media_description_options.mid,
                          *(audio_transport.get()), answer)) {
    return false;
  }
  if (rejected) {
    RTC_LOG(LS_INFO) << "Audio m= section '" << media_description_options.mid
                     << "' being rejected in answer.";
  }
  answer->AddContent(media_description_options.mid, offer_content->type,
                     rejected, std::move(audio_answer));
  return true;
}

首先调用CreateTransportAnswer，然后过滤获得音频编解码器filtered_codecs，并传入到SetCodecsInAnswer。然后调用CreateMediaContentAnswer构建音频的MediaContent。
调用AddContent传入到音频audio_answer，然后返回。